Danny: I restarted asterisk both times after making changes to the sip.conf in response to Eric's suggestions.
I'll send you my original post with sip debug info. I ran sip debug again. The logged info is identical to what I was getting before I changed the codec allow/disallow settings... I think this is an Asterisk behind a NAT configuration problem, but if I could be wrong. Bud On Fri, 2008-12-19 at 16:38 -0600, Danny Nicholas wrote: > Have you done a sip set debug, then sip reload? Do you have a range of > 10000-20000 open in your firewall? Asterisk will "poke out" through 5060 > but has to get a random response back in the 10-20K range (you can narrow > this) > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of obitori junk > Sent: Friday, December 19, 2008 4:28 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem configuring Asterisk as client for > ekiga.net -- NAT problem > > I tried both "disallow all /// allow ulaw" and "allow all". Neither > worked. Ekiga.net is not accepting asterisk as a client. I keep > getting the "600 not acceptable error". What seems odd to me is that > the NAT column shows an "n" for both ekiga.net and jnctn.net, but I have > "nat=yes" for both. Thanks for the suggestion...I'm still open to > ideas. > > Regards, > > Bud Roth > > sip.conf contains: "allow all" > OUTPUT--> > > Name/username Host Dyn Nat ACL Port > Status > ekiga/budzhaus 86.64.162.35 N 5060 OK (102 > ms) > 110 (Unspecified) D 0 > UNKNOWN > 109 (Unspecified) D 0 > UNKNOWN > 108 (Unspecified) D 0 > UNKNOWN > 107 (Unspecified) D 0 > UNKNOWN > 106 (Unspecified) D 0 > UNKNOWN > 105 (Unspecified) D 0 > UNKNOWN > 104 (Unspecified) D 0 > UNKNOWN > 103 (Unspecified) D 0 > UNKNOWN > 102/102 10.1.1.20 D 5061 OK (1 > ms) > 101 (Unspecified) D 0 > UNKNOWN > jnctn/obitori 66.227.100.20 N 5060 OK (22 > ms) > 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 > offline] > [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout: -- > Registration for '[email protected]' timed out, trying again (Attempt > #2) > -- Got SIP response 606 "Not Acceptable" back from 86.64.162.35 > > sip.conf contains: > > disallow all > allow ulaw > > OUTPUT--> > victoria*CLI> sip show peers > Name/username Host Dyn Nat ACL Port > Status > ekiga/budzhaus 86.64.162.35 N 5060 OK (97 > ms) > 110 (Unspecified) D 0 > UNKNOWN > 109 (Unspecified) D 0 > UNKNOWN > 108 (Unspecified) D 0 > UNKNOWN > 107 (Unspecified) D 0 > UNKNOWN > 106 (Unspecified) D 0 > UNKNOWN > 105 (Unspecified) D 0 > UNKNOWN > 104 (Unspecified) D 0 > UNKNOWN > 103 (Unspecified) D 0 > UNKNOWN > 102/102 10.1.1.20 D 5061 OK (1 > ms) > 101 (Unspecified) D 0 > UNKNOWN > jnctn/obitori 66.227.100.20 N 5060 OK (23 > ms) > 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 > offline] > victoria*CLI> > > > On Fri, 2008-12-19 at 15:18 -0600, Eric "ManxPower" Wieling wrote: > > obitori junk wrote: > > > I am experiencing a "606 not Acceptable" error trying to set up an > > > Asterisk server as an ekiga.net client. My server is behind a firewall > > > with NAT routing. I have googled this problem and read about Asterisk > > > feeding its local ip address to ekiga.net. That seems to be my > > > problem. > > > > In my experience "Not Acceptable" errors happen because the two > > endpoints cannot agree on a codec. Try allowing all the codecs in your > > softphone and in Asterisk sip.conf [general] do a disallow=all and an > > allow=ulaw. I suggest you do this in [general] when testing because > > it can sometimes be hard to make sure that a peer/friend/user entry is > > actually matching the incoming call. Once you get it working you can > > refine it. > > > > You could be having a NAT issue too, but I don't think it is related to > > your 606 error. > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
